DSP Implementation of Ofdm Acoustic Modem: Master of Technology in Digital Signal Processing
DSP Implementation of Ofdm Acoustic Modem: Master of Technology in Digital Signal Processing
DSP Implementation of Ofdm Acoustic Modem: Master of Technology in Digital Signal Processing
Master of Technology
In
Digital Signal Processing
By
MADHU.A
Under the guidance of
Prof.G.S.RATH
Prof. G.S.RATH
Dept. of E.C.E
National Institute of Technology
Date:29-05-2008
Rourkela-769008
ACKNOWLEDGEMENTS
First of all, I would like to express my deep sense of respect and gratitude towards
my advisor and guide Prof. G.S.Rath, who has been the guiding force behind this work.
I am greatly indebted to him for his constant encouragement, invaluable advice and for
propelling me further in every aspect of my academic life. His presence and optimism
have provided an invaluable influence on my career and outlook for the future. I
consider it my good fortune to have got an opportunity to work with such a wonderful
person.
Next, I want to express my respects to Prof. S.K.Patra, Prof.G.Panda, Prof.
K.K. Mahapatra, and Dr. S. Meher for teaching me and also helping me how to learn.
They have been great sources of inspiration to me and I thank them from the bottom of
my heart.
I would like to thank all faculty members and staff of the Department of
Electronics and Communication Engineering, N.I.T. Rourkela for their generous help in
various ways for the completion of this thesis.
I would also like to mention the name of Jithendra Kumar Das (Ph.D) for
helping me a lot during the thesis period.
I would like to thank all my friends and especially my classmates for all the
thoughtful and mind stimulating discussions we had, which prompted us to think beyond
the obvious. Ive enjoyed their companionship so much during my stay at NIT, Rourkela.
I am especially indebted to my parents for their love, sacrifice, and support. They
are my first teachers after I came to this world and have set great examples for me about
how to live, study, and work.
Madhu.A
Roll No: 20607021
Dept of ECE, NIT, Rourkela.
3
CONTENTS
Acknowledgements
Contents
ii
Abstract
List of figures
vi
List of tables
viii
Abbreviations
ix
Nomenclature
xi
Chapter1 introduction
1.1 Introduction
1.2 Motivation
10
11
11
12
13
14
Chapter 3 OFDM
3.1 Introduction
16
16
17
18
19
20
21
21
22
25
26
26
3.7.3 Preamable
26
27
3.9 RF modulation
28
28
29
30
30
31
33
3.11.2 Disadvantages
33
35
36
37
38
40
41
42
42
45
5
45
46
51
5.5 conclusion
53
References
54
ABSTRACT
The success of multicarrier modulation in the form of OFDM in radio channels
illuminates a path one could take towards high-rate underwater acoustic communications,
and recently there are intensive investigations on underwater OFDM. Processing power
has increased to a point where orthogonal frequency division multiplexing (OFDM) has
become feasible and economical. Since many wireless communication systems being
developed use OFDM, it is a worthwhile research topic. Some examples of applications
using OFDM include Digital subscriber line (DSL), Digital Audio Broadcasting (DAB),
High definition television (HDTV) broadcasting, IEEE 802.11 (wireless networking
standard).OFDM is a strong candidate and has been suggested or standardized in high
speed communication systems.
In this Thesis in first phase ,we analyzes the factor that affects the OFDM
performance. The performance of OFDM was assessed by using computer simulations
performed using Matlab7.2 .it was simulated under Additive white Gaussian noise
(AWGN) ,Exponential Multipath channel and Carrier frequency offset conditions for
different modulation schemes like binary phase shift keying (BPSK), Quadrature phase
shift keying (QPSK), 16-Quadrature amplitude modulation (16-QAM), 64-Quadrature
amplitude modulation (64-QAM) which are used for achieving high data rates.
In second phase we implement the acoustic OFDM transmitter and receiver design of [4,
5] on a TMS320C6713 DSP board. We analyze the workload and identify the most timeconsuming operations. Based on the workload analysis, we tune the algorithms and
optimize the code to substantially reduce the synchronization time to 0.2 seconds and the
processing time of one OFDM block to 2.7235 seconds on a DSP processor at 225 MHz.
This experimentation provides guidelines on our future work to reduce the per-block
processing time to be less than the block duration of 0.23 seconds for real time operations
LIST OF FIGURES
2.1 Scenario of the mobile UWSN architecture
12
17
18
22
23
24
25
26
27
28
3.8 RF modulation of complex base band OFDM signal, using analog techniques 29
3.10 Addition of a guard period to an OFDM signal
30
3.11 Example of intersymbol interference. The green symbol was transmitted first,
followed by the blue symbol.
32
37
40
41
44
45
47
48
50
51
52
5.7 BER vs. SNR plot for OFDM using BPSK, QPSK, 16-QAM, 64-QAM
52
LIST OF TABLES
48
53
ABBREVIATIONS
AWGN
ADSL
BPSK
CCK
CCS
CDMA
DSP
DAB
DVB
DFT
DSSS
ETSI
FFT
FDM
FEC
HDTV
IEEE
IFFT
IDFT
ISI
ICI
LAN
NTSC
OFDM
QPSK
QAM
SNR
TDM
TDMA
UHF
VLSI
WLAN
10
NOMENCLATURE
A C (t)
c (t)
Carrier frequency
c (t)
ss (t)
Fc
Carrier frequency
FS
Sampling rate
TS
TG
TFFT
Lc
Lp
x (n)
original signal
X (k)
WNkn
Twiddle factors
Subcarrier spacing
NFFT
f1 (n)
f2 (n)
F1 (k)
F2 (k)
CHAPTER 1
INTRODUCTION
12
1.1 INTRODUCTION
The ever increasing demand for very high rate wireless data transmission calls for
technologies which make use of the available electromagnetic resource in the most
intelligent way. Key objectives are spectrum efficiency (bits per second per Hertz),
robustness
against
multipath
propagation,
range,
power
consumption,
and
OFDM. Section 1.4 discusses the contribution in this thesis. At the end, section 1.5
presents thesis outline.
1.2 MOTIVATION
OFDM is the modulation technique used in many new and emerging broadband
communication systems including wireless local area networks (WLANs), high definition
television (HDTV) and 4G systems. To achieve high data rates OFDM is used in wireless
LAN standards like IEEE 802.11a, IEEE 802.11g. The key component in an OFDM
transmitter is an inverse fast Fourier transform (IFFT) and in the receiver, an FFT. The
increasing computational power and performance capabilities of DSPs make them ideal
for the practical implementation of OFDM functions.
The motivation for using OFDM techniques over TDMA techniques is twofold.
First, TDMA limits the total number of users that can be sent efficiently over a channel.
In addition, since the symbol rate of each channel is high, problems with multipath delay
spread invariably occur. In stark contrast, each carrier in an OFDM signal has a very
narrow bandwidth (i.e. 1kHz); thus the resulting symbol rate is low. This results in the
signal having a high degree of tolerance to multipath delay spread, as the delay spread
must be very long to cause significant inter-symbol interference.
1.3 BACKGROUND LITERATURE SURVEY
DAB was the first commercial use of OFDM technology [5]. Development of DAB
started in 1987 and services began in U.K and Sweden in1995. DAB is a replacement for
FM audio broadcasting, by providing high quality digital audio and information services.
OFDM was used for DAB due to its multipath tolerance.
Broadcast systems operate with potentially very long transmission distances (20 100 km). As a result, multipath is a major problem as it causes extensive ghosting of the
transmission. This ghosting causes Inter-Symbol Interference (ISI), blurring the time
domain signal.
For single carrier transmissions the effects of ISI are normally mitigated using
adaptive equalization. This process uses adaptive filtering to approximate the impulse
response of the radio channel. An inverse channel response filter is then used to
recombine the blurred copies of the symbol bits. This process is however complex and
15
slow due to the locking time of the adaptive equalizer. Additionally it becomes increasing
difficult to equalize signals that suffer ISI of more than a couple of symbol periods.
OFDM overcomes the effects of multipath by breaking the signal into many
narrow bandwidth carriers. This results in a low symbol rate reducing the amount of ISI.
In addition to this, a guard period is added to the start of each symbol, removing the
effects of ISI for multipath signals delayed less than the guard period. The high tolerance
to multipath makes OFDM more suited to high data transmissions in terrestrial
environments than single carrier transmissions.
The data throughput of DAB varies from 0.6 - 1.8 Mbps depending on the amount
of Forward Error Correction (FEC) applied. This data payload allows multiple channels
to be broadcast as part of the one transmission ensemble. The number of audio channels
is variable depending on the quality of the audio and the amount of FEC used to protect
the signal. For telephone quality audio (24 kbps) up to 64 audio channels can be
provided, while for CD quality audio (256 kb/s), with maximum protection, three
channels are available.
1.3.2 Digital video broadcasting
The development of the Digital Video Broadcasting (DVB) standards was started in
1993. DVB is a transmission scheme based on the MPEG-2 standard, as a method for
point to multipoint delivery of high quality compressed digital audio and video. It is an
enhanced replacement of the analogue television broadcast standard, as DVB provides a
flexible transmission medium for delivery of video, audio and data services [6]. The
DVB standards specify the delivery mechanism for a wide range of applications,
including satellite TV (DVB-S), cable systems (DVB-C) and terrestrial transmissions
(DVB-T). The physical layer of each of these standards is optimized for the transmission
channel being used. Satellite broadcasts use a single carrier transmission, with QPSK
modulation, which is optimized for this application as a single carrier allows for large
Doppler shifts, and QPSK allows for maximum energy efficiency [7]. This transmission
method is however unsuitable for terrestrial transmissions as multipath severely degrades
the performance of high-speed single carrier transmissions. For this reason, OFDM was
16
used for the terrestrial transmission standard for DVB. The physical layer of the DVB-T
transmission is similar to DAB, in that the OFDM transmission uses a large number of
subcarriers to mitigate the effects of multipath. DVB-T allows for two transmission
modes depending on the number of subcarriers used [8].The major difference between
DAB and DVB-T is the larger bandwidth used and the use of higher modulation schemes
to achieve a higher data throughput. The DVB-T allows for three subcarrier modulation
schemes: QPSK, 16-QAM (Quadrature Amplitude Modulation) and 64- QAM; and a
range of guard period lengths and coding rates. This allows the robustness of the
transmission link to be traded at the expense of link capacity.
1.3.3 Hiperlan2 and IEEE802.11a
Development of the European Hiperlan standard was started in 1995, with the final
standard of HiperLAN2 being defined in June 1999. HiperLAN2 pushes the performance
of WLAN systems, allowing a data rate of up to 54 Mbps [9]. HiperLAN2 uses 48 data
and 4 pilot subcarriers in a 16 MHz channel, with 2 MHz on either side of the signal to
allow out of band roll off. User allocation is achieved by using TDM, and subcarriers are
allocated using a range of modulation schemes, from BPSK up to 64-QAM, depending
on the link quality. Forward Error Correction is used to compensate for frequency
selective fading. IEEE802.11a has the same physical layer as HiperLAN2 with the main
difference between the standard corresponding to the higher-level network protocols
used.HiperLAN2 is used extensively as an example OFDM system in this thesis. Since
the physical layer of HiperLAN2 is very similar to the IEEE802.11a standard these
examples are applicable to both standards.
has demonstrated multicarrier OFDM transmission and reception in air and in a water
tank, where the algorithms in [3, 4] are implemented by Matlab programs in two laptops
[5].
In this Dissertation, in the first phase we simulated the OFDM transmission and reception
algorithms of [3, 4] in MATLAB 7.2.and compared the results. In the second phase we
generated the c code to execute on a TI TMS320C6713 DSP board with a processor
running at 225 MHz. In-wired communications are successfully tested. We analyze the
workload and identify the most time-consuming operations..
1.5 THESIS OUTLINE
19
CHAPTER 2.
UWSN
20
2.1. INTRODUCTION
The earth is a water planet. Currently, there has been a growing interest in monitoring
underwater mediums for scientific exploration, commercial exploitation, and attack
protection. A distributed underwater wireless sensor network (UWSN) is the ideal vehicle
for this monitoring. A scalable UWSN is a good solution for exploring the aquatic
environments. By deploying scalable wireless sensor networks in 3-dimensional
underwater space, each underwater sensor can monitor and find environmental events.
The aqueous systems are also dynamic and processes happen within the water mass as it
disperses within the environment. In a mobile underwater sensor network, the sensor
mobility has two major benefits:
1. Mobile sensors injected in the current in relative large numbers can help to track
changes in the water mass, thus provide 4D (space and time) environmental sampling.
2. Floating sensors can help to form dynamic monitoring coverage and increase system
reusability.
The self-organizing network of mobile sensors produces better supports in sensing,
monitoring, surveillance, scheduling, underwater control, and failing tolerance. Mobile
UWSNs have to use acoustic communications, since radio does not work well in
underwater environments. Due to the unique features of large latency, low bandwidth,
and high error rate, underwater acoustic channels bring much defiance to the protocol
design. Furthermore, the best parts of underwater nodes are mobile due to water currents.
This mobility is another problem to consider in the system design.
21
22
Path loss:
Noise:
Man made noise: This is mainly caused by machinery noise and shipping
activity.
Multi-path:
Delay: The propagation speed in the underwater acoustic channels is five orders
of magnitude lower than in the radio channel. This large propagation delay can
reduce the throughput of the system considerably.
23
Delay variance: The very high delay variance is even more harmful for efficient
protocol design, as it prevents from accurately estimating the round trip time, key
measure for many common communication protocols.
Doppler spread:
Underwater acoustic channels are temporally spatially and variable due to the
characteristics of the transmission medium and physical properties of the environments.
The signal propagation speed in underwater acoustic channel is about 1.5 103 m/sec.
The convenient bandwidth of underwater acoustic channels is limited and dramatically
depends on both transmission range and frequency. The acoustic band under water is
restricted due to absorption.
The bandwidth of underwater acoustic channels working over several kilometers is about
several tens of kbps, whereas short-range systems over several tens of meters can reach at
hundreds of kbps. The path loss, noise, multipath, and Doppler spread affect the
underwater acoustic communication channels. All these factors generate high bit-error
and delay variance.
2.2.3. Distinctions between mobile UWSNs and ground-based sensor networks
A mobile UWSN is very different from any ground-based sensor network in the
following aspects:
Node Mobility: The sensor nodes in ground-based sensor networks are fixed,
though it is possible to implement interactions between these static sensor nodes
and a limit number of mobile nodes. However, the best part of underwater sensor
24
nodes are with low or medium mobility due to water current and other
underwater activities. From experimental observations, underwater objects may
move at the speed of 3-6 kilometers per hour in a typical underwater condition.
2.2.4. Current underwater network systems
An underwater sensor network is a next step forward with respect to existing small-scale
Underwater Acoustic Networks (UANs). UANs are associations of nodes that collect data
using remote telemetry or assuming point-to-point communications. The different
between UANs and underwater sensor networks are the following:
25
CHAPTER 3
OFDM
26
3.1 INTRODUCTION
impulse noise, signal reflections and other impairments. These impairments can impede
the ability to recover the information sent. In addition, as the bandwidth used by a single
carrier system increases, the susceptibility to interference from other continuous signal
sources becomes greater. This type of interference is commonly labeled as carrier wave
(CW) or frequency interference.
3.3 FREQUENCY DIVISION MULTIPLEXING MODULATION SYSTEM
guard bands lower the systems effective information rate when compared to a single
carrier system with similar modulation.
If the FDM system above had been able to use a set of sub carriers that were orthogonal
to each other, a higher level of spectral efficiency could have been achieved. The guard
bands that were necessary to allow individual demodulation of sub carriers in an FDM
system would no longer be necessary. The use of orthogonal sub carriers would allow the
sub carriers spectra to overlap, thus increasing the spectral efficiency. As long as
orthogonality is maintained, it is still possible to recover the individual sub carriers
signals despite their overlapping spectrums. If the dot product of two deterministic
signals is equal to zero, these signals are said to be orthogonal to each other.
Orthogonality can also be viewed from the standpoint of stochastic processes. If two
random processes are uncorrelated, then they are orthogonal. Given the random nature of
signals in a communications system, this probabilistic view of orthogonality provides an
intuitive understanding of the implications of orthogonality in OFDM.
OFDM is implemented in practice using the discrete Fourier transform (DFT). Recall from
signals and systems theory that the sinusoids of the DFT form an orthogonal basis set, and
a signal in the vector space of the DFT can be represented as a linear combination of the
orthogonal sinusoids. One view of the DFT is that the transform essentially correlates its
input signal with each of the sinusoidal basis functions. If the input signal has some energy
at a certain frequency, there will be a peak in the correlation of the input signal and the
basis sinusoid that is at that corresponding frequency. This transform is used at the OFDM
transmitter to map an input signal onto a set of orthogonal sub carriers, i.e., the orthogonal
basis functions of the DFT. Similarly, the transform is used again at the OFDM receiver to
process the received sub carriers. The signals from the sub carriers are then combined to
form an estimate of the source signal from the transmitter. The orthogonal and uncorrelated
nature of the sub carriers is exploited in OFDM with powerful results. Since the basis
functions of the DFT are uncorrelated, the correlation performed in the DFT for a given sub
carrier only sees energy for that corresponding sub carrier. The energy from other sub
29
carriers does not contribute because it is uncorrelated. This separation of signal energy is
the reason that the OFDM sub carriers spectrums can overlap without causing interference.
(3.1)
The real signal is the real part of sc (t). Ac (t) and c (t), the amplitude and phase of the carrier, can vary
on a symbol by symbol basis. The values of the parameters are constant over the symbol duration period
t. OFDM consists of many carriers. Thus the complex signal Ss(t) is represented by:
1 N 1
ss (t) = A N (t)e j[ n t +n (t )]
N n =0
(3.2)
Where
This is of course a continuous signal. If we consider the waveforms of each component of
the signal over one symbol period, then the variables Ac (t) and c (t) take on fixed
values, which depend on the frequency of that particular carrier, and so can be rewritten:
n (t) = n
A n (t) = A n
If the signal is sampled using a sampling frequency of 1/T(48kHz), then the resulting
signal is represented by:
ss (kT) =
1 N 1
A n e[ j( 0 + n)kT +n ]
N n =0
30
(3.3)
At this point, we have restricted the time over which we analyze the signal to N(1024)
samples. It is convenient to sample over the period of one data symbol. Thus we have a
relationship: t=NT If we now simplify equation 3.3, without a loss of generality by letting
0=0, then the signal becomes:
1 N 1
s s (kT) = A n e jn e j(n)kT
N N =0
(3.4)
Now equation 3.4 can be compared with the general form of the inverse Fourier
transform:
1
g (kT ) =
N
In Equation 3.4 the function
A n e j n
2
kn
n
N
G
(
)e
NT
n =0
N 1
(3.5)
sampled frequency domain, and s (kT) is the time domain representation. Eqns.4 and 5
are equivalent if:
This is the same condition that was required for orthogonality Thus, one consequence of
maintaining orthogonality is that the OFDM signal can be defined by using Fourier
transform procedures.
3.6 OFDM GENERATION AND RECEPTION
OFDM signals are typically generated digitally due to the difficulty in creating large
banks of phase locks oscillators and receivers in the analog domain. Fig 3.3 shows the
block diagram of a typical OFDM transceiver [15]. The transmitter section converts
digital data to be transmitted, into a mapping of subcarrier amplitude and phase. It then
transforms this spectral representation of the data into the time domain using an Inverse
Discrete Fourier Transform (IDFT). The Inverse Fast Fourier Transform (IFFT) performs
the same operations as an IDFT, except that it is much more computationally efficiency,
and so is used in all practical systems. In order to transmit the OFDM signal the
calculated time domain signal is then mixed up to the required frequency.
31
The receiver performs the reverse operation of the transmitter, mixing the RF signal to
base band for processing, then using a Fast Fourier Transform (FFT) to analyze the signal
in the frequency domain [16]. The amplitude and phase of the sub carriers is then picked
out and converted back to digital data. The IFFT and the FFT are complementary
function and the most appropriate term depends on whether the signal is being received
or generated. In cases where the signal is independent of this distinction then the term
FFT and IFFT is used interchangeably.
32
Modulation
One way to communicate a message signal whose frequency spectrum does not fall
within that fixed frequency range, or one that is otherwise unsuitable for the channel, is to
change a transmittable signal according to the information in the message signal. This
alteration is called modulation, and it is the modulated signal that is transmitted. The
receiver then recovers the original signal through a process called demodulation.
Modulation is a process by which a carrier signal is altered according to information in a
message signal. The carrier frequency, denoted Fc, is the frequency of the carrier signal.
33
The sampling rate, Fs, is the rate at which the message signal is sampled during the
simulation. The frequency of the carrier signal is usually much greater than the highest
frequency of the input message signal. The Nyquist sampling theorem requires that the
simulation sampling rate Fs be greater than two times the sum of the carrier frequency
and the highest frequency of the modulated signal, in order for the demodulator to
recover the message correctly.
For a given modulation technique, two ways to simulate modulation techniques are called
baseband and pass band. Baseband simulation requires less computation. In this thesis,
baseband simulation will be used.
In this method the amplitude of the carrier assumes one of the two amplitudes dependent
on the logic states of the input bit stream. A typical output waveform of an ASK
modulation is shown in Fig3.4.
b) Frequency Shift Key (FSK) Modulation
In this method the frequency of the carrier is changed to two different frequencies
depending on the logic state of the input bit stream. The typical output waveform of an
34
FSK is shown in Fig 3.5. Notice that logic high causes the centre frequency to increase to
a maximum and a logic low causes the centre frequency to decrease to a minimum.
With this method the phase of the carrier changes between different phases determined
by the logic states of the input bit stream. There are several different types of Phase Shift
Key (PSK) modulators. These are:
1 Two-phase (2 PSK)
2 Four-phase (4 PSK)
3 Eight-phase (8 PSK)
4 Sixteen-phase (16 PSK) etc.
d) Quadrature Amplitude Modulation (QAM)
QAM is a method for sending two separate (and uniquely different) channels of
information. The carrier is shifted to create two carriers namely the sine and cosine
versions. The outputs of both modulators are algebraically summed and the result of
which is a single signal to be transmitted, containing the In-phase (I) and Quadrature (Q)
information. The set of possible combinations of amplitudes is a pattern of dots known as
a QAM constellation.
Once each subcarrier has been allocated bits for transmission, they are mapped using a
35
-6
-4
-2
Subcarrier modulation can be implemented using a lookup table, making it very efficient
to implement. In the receiver, mapping the received IQ vector back to the data word
performs sub carrier demodulation.
The serial signal is transformed to parallel alter the modulation using a reshape block.
36
In each OFDM symbol, four of the sub carriers are dedicated to pilot signals in order to
make the coherent detection robust against frequency offsets and phase noise. These pilot
signals shall be placed equally in sub carriers ..
As explained, four pilots are inserted in each OFDM symbol. Pilots . Two fifty six pilots
are inserted in each OFDM symbol in the required subcarriers. The insertion is achieved
with a matrix concatenation block, pilots are inserted in its proper place in each symbol.
3.7.3 Preamble
The preamble is used to detect the start of the packet and to synchronize the receiver as
well. The OFDM symbols should be packed into frames before being sent. A preamble is
added at the beginning of each frame.[23] It helps the receiver to estimate phase and
amplitude errors, thereby allowing it to correct the received signal. In the simulation
37
exposed in this work a preamble consisting on two long training symbols, like the
following one, has been used:
L[-256,255}= {..1, 1, -1, -1, 1, 1, -1, 1, -1, 1, 1, 1, 1, 1, 1, -1,-1, 1, 1, -1,1, -1, 1, 1, 1, 1,
0, 1, -1, -1, 1, 1, -1, 1, -1, 1, -1, -1, -1, -1, -1, 1,1, -1, -1, 1,-1, 1, -1, 1, 1, 1, 1}
A long OFDM training symbol consists of 512 sub carriers (including a zero value at
DC).
After the subcarrier modulation stage each of the data sub carriers is set to amplitude and
phase based on the data being sent and the modulation scheme. All unused sub carriers
are set to zero. This sets up the OFDM signal in the frequency domain. An IFFT is then
used to convert this signal to the time domain, allowing it to be transmitted. Fig 3.7
shows the IFFT section of the OFDM transmitter. In the frequency domain, before
applying the IFFT, each of the discrete samples of the IFFT corresponds to an individual
sub carrier. Most of the sub carriers are modulated with data. The outer sub carriers are
unmodulated and set to zero amplitude. These zero sub carriers provide a frequency
guard band before the nyquist frequency and effectively act as an interpolation of the
38
signal and allows for a realistic roll off in the analog anti-aliasing reconstruction filters.
3.9 RF modulation
The output of the OFDM modulator generates a base band signal, which must be mixed
up to the required transmission frequency. This can be implemented using analog
techniques as shown in Fig 3.8 or using a Digital up Converter as shown in Fig 3.9.
Fig 3.8 RF modulation of complex base band OFDM signal, using analog techniques
For a given system bandwidth the symbol rate for an OFDM signal is much lower than a
single carrier transmission scheme. For example for a single carrier BPSK modulation,
the symbol rate corresponds to the bit rate of the transmission. However for OFDM the
system bandwidth is broken up into NC sub carriers, resulting in a symbol rate that is NC
times lower than the single carrier transmission. This low symbol rate makes OFDM
naturally resistant to effects of Inter-Symbol Interference (ISI) caused by multipath
propagation. Multipath propagation is caused by the radio transmission signal reflecting
off objects in the propagation environment, such as walls, buildings, mountains, etc.
These multiple signals arrive at the receiver at different times due to the
transmission distances being different. This spreads the symbol boundaries causing
energy leakage between them. The effect of ISI on an OFDM signal can be further
39
improved by the addition of a guard period to the start of each symbol. This guard period
is a cyclic copy that extends the length of the symbol waveform. Each sub carrier, in the
data section of the symbol, (i.e. the OFDM symbol with no guard period added, which is
equal to the length of the IFFT size used to generate the signal) has an integer number of
cycles. Because of this, placing copies of the symbol end-to-end results in a continuous
signal, with no discontinuities at the joins. Thus by copying the end of a symbol and
appending this to the start results in a longer symbol time. Fig 3.10 shows the insertion of
a guard period.
The total length of the symbol is TS=TG + TFFT, where Ts is the total length of the symbol
in samples, TG is the length of the guard period in samples, and TFFT is the size of the
IFFT used to generate the OFDM signal. In addition to protecting the OFDM from ISI,
the guard period also provides protection against time-offset errors in the receiver. The
effects of multipath propagation and how cyclic prefix reduces the inter symbol
interference is discussed in detail in chapter4.
3.10.1 Protection against time offset
To decode the OFDM signal the receiver has to take the FFT of each received symbol, to
work out the phase and amplitude of the sub carriers. For an OFDM system that has the
same sample rate for both the transmitter and receiver, it must use The same FFT size at
both the receiver and transmitted signal in order to maintain sub carrier orthogonality.
40
Each received symbol has TG + TFFT samples due to the added guard period. The
receiver only needs TFFT samples of the received symbol to decode the signal [18]. The
remaining TG samples are redundant and are not needed. For an ideal channel with no
delay spread the receiver can pick any time offset, up to the length of the guard period,
and still get the correct number of samples, without crossing a symbol boundary. Because
of the cyclic nature of the guard period changing the time offset simply results in a phase
rotation of all the sub carriers in the signal. The amount of this phase rotation is
proportional to the sub carrier frequency, with a sub carrier at the nyquist frequency
changing by 180 for each sample time offset. Provided the time offset is held constant
from symbol to symbol, the phase rotation due to a time offset can be removed out as part
of the channel equalization [19]. In multipath environments ISI reduces the effective
length of the guard period leading to a corresponding reduction in the allowable time
offset error. The addition of guard period removes most of the effects of ISI. However in
practice, multipath components tend to decay slowly with time, resulting in some ISI
even when a relatively long guard period is used.
3.10.2 Guard period overhead and sub carrier spacing
Adding a guard period lowers the symbol rate, however it does not affect the sub carrier
spacing seen by the receiver. The sub carrier spacing is determined by the sample rate
and the FFT size used to analyze the received signal.
f =
FS
N FFT
(3.6)
In Equation (3.6), f is the sub carrier spacing in Hz, Fs is the sample rate in Hz, and
NFFT is the size of the FFT. The guard period adds time overhead, decreasing the overall
spectral efficiency of the system.
Assume that the time span of the channel is Lc samples long. Instead of a single carrier
with a data rate of R symbols/ second, an OFDM system has N subcarriers, each with a
data rate of R/N symbols/second. Because the data rate is reduced by a factor of N, the
41
Fig 3.11 Example of intersymbol interference. The green symbol was transmitted first,
The guard interval is not used in practical systems because it does not prevent an OFDM
symbol from interfering with itself. This type of interference is called intrasymbol
interference [21]. The solution to the problem of intrasymbol interference involves a
discrete-time property. Recall that in continuous-time, a convolution in time is equivalent
to a multiplication in the frequency-domain. This property is true in discrete-time only if
the signals are of infinite length or if at least one of the signals is periodic over the range
of the convolution. It is not practical to have an infinite-length OFDM symbol, however,
it is possible to make the OFDM symbol appear periodic.
This periodic form is achieved by replacing the guard interval with something
known as a cyclic prefix of length Lp samples. The cyclic prefix is a replica of the last Lp
42
samples of the OFDM symbol where Lp > Lc. Since it contains redundant information,
the cyclic prefix is discarded at the receiver. Like the case of the guard interval, this step
removes the effects of intersymbol interference. Because of the way in which the cyclic
prefix was formed, the cyclically-extended OFDM symbol now appears periodic when
convolved with the channel. An important result is that the effect of the channel becomes
multiplicative.
In a digital communications system, the symbols that arrive at the receiver have
been convolved with the time domain channel impulse response of Length Lc samples.
Thus, the effect of the channel is convolution. In order to undo the effects of the channel,
another convolution must be performed at the receiver using a time domain filter known
as an equalizer. The length of the equalizer needs to be on the order of the time span of
the channel. The equalizer processes symbols in order to adapt its response in an attempt
to remove the effects of the channel. Such an equalizer can be expensive to implement in
hardware and often requires a large number of symbols in order to adapt its response to a
good setting. In OFDM, the time-domain signal is still convolved with the channel
response [22]. However, the data will ultimately be transformed back into the frequencydomain by the FFT in the receiver. Because of the periodic nature of the cyclicallyextended OFDM symbol, this time-domain convolution will result in the multiplication of
the spectrum of the OFDM signal (i.e., the frequency- domain constellation points) with
the frequency response of the channel.
The result is that each sub carriers symbol will be multiplied by a complex
number equal to the channels frequency response at that sub carriers frequency. Each
received sub carrier experiences a complex gain (amplitude and phase distortion) due to
the channel. In order to undo these effects, a frequency- domain equalizer is employed.
Such an equalizer is much simpler than a time-domain equalizer. The frequency domain
equalizer consists of a single complex multiplication for each sub carrier. For the simple
case of no noise, the ideal value of the equalizers response is the inverse of the channels
frequency response [24].
43
1 The OFDM signal has a noise like amplitude with a very large dynamic range,
therefore it requires RF power amplifiers with a high peak to average power ratio.
2 It is more sensitive to carrier frequency offset and drift than single carrier systems are
due to leakage of the DFT.
44
CHAPTER 4
Digital Signal processing (DSP) is one of the fastest growing fields of technology and
computer science in the world. In today's world almost everyone uses DSPs in their
everyday life but, unlike PC users, almost no one knows that he/she is using DSPs.
Digital Signal Processors are special purpose microprocessors used in all kind of
electronic products, from mobile phones, modems and CD players to the automotive
industry; medical imaging systems to the electronic battlefield and from dishwashers to
satellites.[17] DSP is all about analysing and processing real-world or analogue signals,
i.e. the kind of signals that humans interact with, for example speech. These signals are
converted to a format that computers can understand (digital) and, once this has
happened, process. The following diagram shows the typical component parts of a DSP
system.
In order to process analog signals with digital computers they must first be converted to
digital signals using analog to digital converters. Similarly, the digital signals must be
converted back to analog ones for them to be used outside the computer.
46
There are many reasons why we process these analog signals in the digital world.
Traditional signal processing was achieved by using analogue components such as
resistors, capacitors and inductors. However, the inherent tolerance associated with this
components, temperature and voltage changes and mechanical vibrations can
dramatically affect the effectiveness of analogue circuitry. On the other hand, DSP is
inherently stable, reliable and repeatable.With DSP it is easy to chance, correct or update
applications. Additionally, DSP reduces noise susceptibility, chip count, development
time, cost and power consumption.
DSP has many unique properties. It is a Super Mathematician thanks to its arithmetic
logic units and its optimized multipliers. DSPs do really well in application where the
data to be processed is arriving in a continuous flow, often referred to as a stream. It uses
almost no power compared to a PC microprocessor. Next, some features that make DSP
different from other microprocessors are going to be described:
1. High
speed
arithmetic:
Most
DSP
operations
require
additions
and
47
simultaneously. For this reason DSP processors usually support multiple memory
accesses in the same instruction cycle.
4. Digital Signal Processors also have the advantage of consuming less power and
being relatively cheap.
The DSP architecture is a well defined but quite complex hardware structure that needs
much time to be explained in detail. An overview of this architecture is going to be
exposed here in order to make it as much understandable as possible.
The TMS320C6000 family of processors from the company Texas Instruments is
designed to meet the real-time requirements of high performance digital signal
processing. With a performance of up to 2000 million instructions per second (MIPS) at
250 MHz and a complete set of development tools, the TMS320C6000 DSPs offer cost
effective solutions to higher-performance DSP programming challenges.
The TMS320C6000 DSPs give the system architects unlimited possibilities to
differentiate their products. High performance, easy use, and affordable pricing make the
TMS320c6000 platform the ideal solution for a large number of applications
(multichannel multifunction applications such as: pooled modems, wireless local loop
base stations, multichannel telephony systems, etc). First of all, a DSP device must be
considered as a specific microprocessor whose components have been linked in a clever
way to process faster.
The TMS320C6xxx family are processors currently running at a clock speed of up to
300MHz (225MHz in the TMS320C6713 case). The C62xx processors are fixed-point
processors whereas the C67xx are floating-point processors. These refer to the format
used to store and manipulate numbers withing the devices. Figure 24 shows the main
components of the TMS320C6000 DSP under a block diagram form.
It is composed of:
1. External Memory Interface (EMIF) to access external data at the specified
address.
48
2. Memory, which is the internal memory where a set of instructions and data
values can be stored (FFT algorithm for example).
3. Peripherals are the possible connectable devices that can be associated with the
DSP (DMA/EDMA, Serial port, Timer/Counter).
4. Internal buses; they allow the components to quickly communicate together
differentiating addresses and data.
5. CPU, which is the most important component since it performs all the operation.
The TMS320C67 DSPs (including the TMS320C6713 device) compose the floating
point DSP generation in the TMS320C6000 DSP platform. The TMS320C6713 (C6713)
device is based on the high-performance, advanced VelociTI very-long-instruction-word
(VLIW) architecture developed by Texas Instruments (TI), making this DSP an excellent
choice for multichannel and multifunction applications.
The DSK features the TMS320C6713 DSP, a 225 MHz device delivering up to 1800
million instructions per second (MIPs) and 1350 MFLOPS. This DSP generation is
designed for applications that require high precision accuracy. The C6713 is based on the
TMS320C6000 DSP platform designed to fit the needs of high-performing high-precision
applications such as pro-audio, medical and diagnostic. Other hardware features of the
TMS320C6713 DSK board include:
49
50
We taken audio signal as input to the system,To sample audio signals, one of the
multichannel buffered serial ports (McBSPs) is configured to connect the AIC23 codec.
The audio data is transferred between the codec and the internal L2 memory through the
enhanced direct memory access (EDMA) channel. To save the raw audio data from the
AIC23 codec continuously, the commonly-known double-buffering method is used.
When one of the two buffers is filled, a DMA interrupt is initiated and the data is passed
to the interrupt service routine (ISR) and then is processed. At the same time, the codec
keeps sampling and saves data into the other buffer. So data sampling and processing can
be done simultaneously and no incoming signals are missed even if the DSP is processing
previously received data.
Processing steps:
52
4.4.2 CHANNELNOISE:
4.4.3 RECEIVER:
1. Qpsk Demodultion takes place to extract the binay symols from carrier.(techinge
applied PLL.
53
2. Autocorrelation and cross correlation peaks of long preamble of two period sequences
with 60% cyclic prefix decide the starting symbol of OFDM frame.This technique
provides us coarse frequency tuning for 50% overlap of sub carries.
3. FFT will be calculated to covert time domain symbols into frequency domain where
already effected by Phase Noise and frequency noise
4. CFO estimation done by taking FFT of Kn sub carriers where minargJ(n) minimum
then ICI would be greatly reduced.
5. One dimensional channel estimation done by calculating the FFT of K/4 Pilot sub
carriers.
6. Corrupted qpsk symbols extracted by quantization technique.
7. Demapping (mapped symbols converted back into bits).
8. Vertebra decoder(hard decision algorithm) decodes the corrupted data and provides the
output to the speaker.
54
CHAPTER 5
SIMULATION RESULTS
55
5.1 INTRODUCTION
An OFDM system was modeled using Matlab to allow various parameters of the system
to be varied and tested. The aim of doing the simulations was to measure the performance
of OFDM under AWGN, Multipath(real world exponential channel considered) channels,
Carrier frequency offset conditions, for different modulation schemes like BPSK, QPSK,
16-QAM, 64-QAM used in IEEE 802.11a wireless LAN standard. and in CCS for
Modem design we considered only QPSK modulation in wired environment.
Following this introduction, section 5.2 discusses model used in simulation, steps
in OFDM simulation and parameters used for the Modem design. Section 5.3 presents
one important block in receiver Frame Synchronization in detail. Section 5.4 provides the
simulation results of OFDM system for different channel schemes.
5.2 SIMULATION MODEL
The OFDM system that was simulated using matlab for the model shown in Fig 5.1.
56
Parameter
value
modulations used
BPSK,QPSK,16-QAM,64-QAM
FFT size
1024
128 samples
In order to properly demodulate the transmitted data, the start of each OFDM frame
needs to be found with reasonably accuracy. This is the task of the OFDM frame
synchronization subsystem. The OFDM frame synchronization subsystem ignores input
before the preamble comes in and then aligns the input directly after the preamble on
frame boundaries..
57
Since every symbol is the same length, if the start of the OFDM symbols can be found,
then the decoding of the symbols can be properly performed.
Theory
1.2
amplitude
0.8
0.6
0.4
0.2
50
100
received samples
58
150
In above Equation, L is the length of one short symbol, r is the whole transmitted
preamble, and N is the length of the OFDM data (512 samples). The result of Equation 8
is rather noisy. It is then smoothed with a moving average filter as shown in below
Equation.
In Equation The spikes in Figure 5.3 are generated by Equation 5.4. In this equation, the
whole preamble, r, is cross correlated with one short symbol, s. S is the number of FFT
samples, as in Equation 5.2 and M is the number of short symbols that the cross
correlation is averaged over.
The start of the OFDM symbol can then be found by timing off of the last large spike
inside the flat part of the dome.
In this way for frame synchronization we perform several tests under different channels
and for different modulations, ploted the BER Vs SNR and MSE Vs SNR graphs.
59
0.9
0.8
0.7
amplitude
0.6
0.5
0.4
0.3
0.2
0.1
0
500
1000
received short preamble
1500
10
AWGN channel
Multipath DS 50ms
Multipath DS 50ms&Freq.offset 0.2Hz
-1
10
-2
10
-3
10
-4
10
8
SNR(db)
10
12
14
60
10
10
-2
MSE
10
-3
10
-4
10
10
12
SNR(db)
14
16
18
20
BER vs SNR
10
BPSK
QPSK
16-QAM
64-QAM
-1
BER
10
-2
10
-3
10
-4
10
10
15
SNR
20
25
30
Fig.5.7 BER vs. SNR plot for OFDM using BPSK, QPSK, 16-QAM, 64-QAM
61
FUNCTIONS
COUNT
EXECUTION TIME
Convlution
6751552
mul_sum
10214
sum_seq
32764
Synchronization
3637253
Complex_conv
38546754
CFO_estimate
34080023
FFT
23
884811
VITERBI
1367923
Block_Processing
47262925
TOTAL
473578235
PCM:
Speech input 16 bit converted into binary and transmitted though cable successfully
received with out noise
Maximum input frequency=1 kHz
62
DPCM:
Applied mu law for bit compression and used LPC coder to provide prediction error
minimum to get less coded bits (reduced from 16 to 8) observed signal clearly .
Maximum input frequency=1 kHz
Sampling frequency =16 kHz
Single carrier Modulation:
BPSK,QPSK:
Successfully tested, components band pass filter (16 kHz), multiplier,lowpass
filter(4kHz)(with 30th order and FIR filter with Kaiser window),decimator
Maximum input frequency=1 kHz
Sampling frequency =8 kHz
Filtering operations done by overlap add FFT method to improving the speed factor than
convolution method
Multi carrier Modulation:
Tested Successfully in SIMULINK and that model Embedded into DSP target for
execution. Frequencies separated by band pas filters (IIR design with 30 order)
Input frequencies=1 kHz, 2 kHz, 3 kHz
Sampling frequency =8 kHz
OFDM:
The code we generated Loaded onto the 2 DSKS (Transmitter, Receiver) And When
tested we were hearing more noise than signal. We tried with all possible conditions but
could not improve the system performance. and at last we measured all the subroutines
code length and their execution time by using CCS profiler And we found that for one
block OFDM our processor taking approxly 2.7235 sec Actually which could finish in
OFDM symbol duration which is(0.2298sec) We tried our best at optimizing the code and
utilize the resources to the maximum extent.
63
CONCLUSION
LIMITATIONS
1.We could not test the system in real time in air to air and in under water communication
environments.
2.we did not consider Doppler spread in simulation.
FUTURE WORK
1.the current implementation does not meet the real-time operation requirements yet.
2.In the future, we can motivated to pursue a hybrid DSP/FPGA-based solution to
construct a real-time OFDM modem.
64
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[1] Hai Yan, Shengli Zhou, Zhijie Jerry Shi, and Baosheng Li DSP Implementaion of
OFDM Acoustic Modem in WUWNet07, September 14, 2007,
[2] M. Stojanovic, Low complexity OFDM detector for underwater channels, in Proc.
of OCEANS, Sept. 2006.
[6] IEEE Standard 802.11a. IEEE Standards Board. 1999 Edition. Piscatawa, USA.
[7] IEEE Standard 802.11a White Paper. IEEE Standards Board.
[8] Digital Signal Processing and Applications with the C6713 and C6416 DSK
By
65
[12] Robert Chang, Orthogonal frequency division multiplexing, US. Patent 3,488445,
filed November 14, 1966, issued January 6, 1970.
[13] Robert Chang, Synthesis of Band-Limited Orthogonal Signals for Multichannel
Data Transmission, the Bell System Technical Journal, December 1966, pp. 1775 -1796.
[14] S. B. Weinstein, Paul M. Ebert, Data Transmission by Frequency-Division
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[15] Louis Thibault, Minh Thien Le, Performance Evaluation of COFDM for Digital
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[16] Ulrich Reimers, Digital Video Broadcasting, IEEE Communications Magazine,
June 1998, pp. 104 - 110
[17] ETSI EN 300 421, Framing Structure, channel coding and modulation for 11/12
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[18] ETSI ETS 300 744, Framing Structure, Channel Coding and Modulation for Digital
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[19] M. Johnsson, HiperLAN/2 The Broadband Radio Transmission Technology
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Online: http://www.hiperlan2.com/site/specific/specmain/specwh.htm, 1999
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66